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avfilter/af_silenceremove: switch to activate
Do full rewrite, new code is much faster for typical filter usages. Also previous code was hard to follow and basically very-hard to maintain.
This commit is contained in:
Paul B Mahol
parent
82278e8749
commit
3235de4883
@ -6431,7 +6431,7 @@ Set the count for trimming silence from the end of audio.
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To remove silence from the middle of a file, specify a @var{stop_periods}
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that is negative. This value is then treated as a positive value and is
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used to indicate the effect should restart processing as specified by
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@var{start_periods}, making it suitable for removing periods of silence
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@var{stop_periods}, making it suitable for removing periods of silence
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in the middle of the audio.
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Default value is @code{0}.
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File diff suppressed because it is too large
Load Diff
293
libavfilter/silenceremove_template.c
Normal file
293
libavfilter/silenceremove_template.c
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@ -0,0 +1,293 @@
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/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#undef ftype
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#undef FABS
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#undef FMAX
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#undef SAMPLE_FORMAT
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#undef SQRT
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#undef ZERO
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#if DEPTH == 32
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#define SAMPLE_FORMAT flt
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#define SQRT sqrtf
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#define FMAX fmaxf
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#define FABS fabsf
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#define ftype float
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#define ZERO 0.f
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#else
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#define SAMPLE_FORMAT dbl
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#define SQRT sqrt
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#define FMAX fmax
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#define FABS fabs
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#define ftype double
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#define ZERO 0.0
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#endif
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#define fn3(a,b) a##_##b
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#define fn2(a,b) fn3(a,b)
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#define fn(a) fn2(a, SAMPLE_FORMAT)
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static void fn(flush)(ftype *dst, const ftype *src, int src_pos,
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int nb_channels, int count, int src_nb_samples,
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int *out_nb_samples)
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{
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int oidx, out_count = count;
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int sidx = src_pos;
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if (count <= 0)
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return;
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oidx = *out_nb_samples + out_count - 1;
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*out_nb_samples += out_count;
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while (out_count-- > 0) {
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const int spos = sidx * nb_channels;
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const int opos = oidx * nb_channels;
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for (int ch = 0; ch < nb_channels; ch++)
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dst[opos + ch] = src[spos + ch];
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oidx--;
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sidx--;
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if (sidx < 0)
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sidx = src_nb_samples - 1;
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}
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}
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static void fn(queue_sample)(AVFilterContext *ctx,
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const ftype *src,
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ftype *queue,
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int *queue_pos,
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int *queue_size,
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int *window_pos,
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int *window_size,
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const int nb_channels,
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const int nb_samples,
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const int window_nb_samples)
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{
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const int pos = *queue_pos * nb_channels;
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for (int ch = 0; ch < nb_channels; ch++)
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queue[pos + ch] = src[ch];
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(*queue_pos)++;
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if (*queue_pos >= nb_samples)
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*queue_pos = 0;
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if (*queue_size < nb_samples)
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(*queue_size)++;
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if (*window_size < window_nb_samples)
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(*window_size)++;
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(*window_pos)++;
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if (*window_pos >= window_nb_samples)
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*window_pos = 0;
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}
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static ftype fn(compute_peak)(ftype *cache, ftype sample, ftype wsample,
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int window_size)
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{
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ftype r;
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cache[0] += FABS(sample);
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cache[0] -= FABS(wsample);
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cache[0] = r = FMAX(cache[0], ZERO);
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return r / window_size;
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}
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static ftype fn(compute_rms)(ftype *cache, ftype sample, ftype wsample,
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int window_size)
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{
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ftype r;
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cache[0] += sample * sample;
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cache[0] -= wsample * wsample;
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cache[0] = r = FMAX(cache[0], ZERO);
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return SQRT(r / window_size);
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}
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static void fn(filter_start)(AVFilterContext *ctx,
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const ftype *src, ftype *dst,
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int *nb_out_samples,
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const int nb_channels)
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{
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SilenceRemoveContext *s = ctx->priv;
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const int start_periods = s->start_periods;
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int out_nb_samples = *nb_out_samples;
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const int start_window_nb_samples = s->start_window->nb_samples;
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const int start_nb_samples = s->start_queuef->nb_samples;
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const int start_wpos = s->start_window_pos * nb_channels;
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const int start_pos = s->start_queue_pos * nb_channels;
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ftype *startw = (ftype *)s->start_window->data[0];
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ftype *start = (ftype *)s->start_queuef->data[0];
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const ftype start_threshold = s->start_threshold;
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const int start_mode = s->start_mode;
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int start_thres = (start_mode == T_ANY) ? 0 : 1;
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const int start_duration = s->start_duration;
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ftype *start_cache = (ftype *)s->start_cache;
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const int start_silence = s->start_silence;
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fn(queue_sample)(ctx, src, start,
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&s->start_queue_pos,
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&s->start_queue_size,
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&s->start_window_pos,
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&s->start_window_size,
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nb_channels,
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start_nb_samples,
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start_window_nb_samples);
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for (int ch = 0; ch < nb_channels; ch++) {
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ftype start_sample = start[start_pos + ch];
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ftype start_ow = startw[start_wpos + ch];
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ftype tstart;
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tstart = fn(s->compute)(start_cache + ch,
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start_sample,
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start_ow,
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s->start_window_size);
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startw[start_wpos + ch] = start_sample;
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if (start_mode == T_ANY) {
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start_thres |= tstart > start_threshold;
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} else {
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start_thres &= tstart > start_threshold;
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}
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}
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if (s->start_found_periods >= 0) {
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if (start_silence > 0) {
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s->start_silence_count++;
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if (s->start_silence_count > start_silence)
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s->start_silence_count = start_silence;
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}
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s->start_sample_count += start_thres;
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}
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if (s->start_sample_count > start_duration) {
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s->start_found_periods++;
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if (s->start_found_periods >= start_periods) {
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fn(flush)(dst, start, s->start_queue_pos, nb_channels,
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s->start_silence_count, start_nb_samples,
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&out_nb_samples);
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s->start_silence_count = 0;
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s->start_found_periods = -1;
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}
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s->start_sample_count = 0;
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}
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if (s->start_found_periods < 0) {
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const int dst_pos = out_nb_samples * nb_channels;
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for (int ch = 0; ch < nb_channels; ch++)
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dst[dst_pos + ch] = start[start_pos + ch];
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out_nb_samples++;
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}
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*nb_out_samples = out_nb_samples;
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}
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static void fn(filter_stop)(AVFilterContext *ctx,
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const ftype *src, ftype *dst,
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int *nb_out_samples,
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const int nb_channels)
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{
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SilenceRemoveContext *s = ctx->priv;
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const int stop_periods = s->stop_periods;
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int out_nb_samples = *nb_out_samples;
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const int stop_window_nb_samples = s->stop_window->nb_samples;
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const int stop_nb_samples = s->stop_queuef->nb_samples;
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const int stop_wpos = s->stop_window_pos * nb_channels;
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const int stop_pos = s->stop_queue_pos * nb_channels;
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ftype *stopw = (ftype *)s->stop_window->data[0];
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const ftype stop_threshold = s->stop_threshold;
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ftype *stop = (ftype *)s->stop_queuef->data[0];
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const int stop_mode = s->stop_mode;
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int stop_thres = (stop_mode == T_ANY) ? 0 : 1;
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const int stop_duration = s->stop_duration;
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ftype *stop_cache = (ftype *)s->stop_cache;
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const int stop_silence = s->stop_silence;
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const int restart = s->restart;
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fn(queue_sample)(ctx, src, stop,
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&s->stop_queue_pos,
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&s->stop_queue_size,
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&s->stop_window_pos,
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&s->stop_window_size,
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nb_channels,
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stop_nb_samples,
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stop_window_nb_samples);
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for (int ch = 0; ch < nb_channels; ch++) {
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ftype stop_sample = stop[stop_pos + ch];
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ftype stop_ow = stopw[stop_wpos + ch];
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ftype tstop;
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tstop = fn(s->compute)(stop_cache + ch,
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stop_sample,
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stop_ow,
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s->stop_window_size);
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stopw[stop_wpos + ch] = stop_sample;
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if (stop_mode == T_ANY) {
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stop_thres |= tstop <= stop_threshold;
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} else {
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stop_thres &= tstop <= stop_threshold;
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}
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}
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s->found_nonsilence = FFMAX(s->found_nonsilence, !stop_thres);
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if (restart && !stop_thres)
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s->stop_found_periods = 0;
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if (s->stop_found_periods >= 0) {
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if (s->found_nonsilence) {
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s->stop_sample_count += stop_thres;
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s->stop_sample_count *= stop_thres;
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}
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} else if (s->stop_silence_count > 0) {
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const int dst_pos = out_nb_samples * nb_channels;
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for (int ch = 0; ch < nb_channels; ch++)
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dst[dst_pos + ch] = stop[stop_pos + ch];
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s->stop_silence_count--;
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out_nb_samples++;
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}
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if (s->stop_sample_count > stop_duration) {
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s->stop_found_periods++;
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if (s->stop_found_periods >= stop_periods) {
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s->stop_found_periods = -1;
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s->stop_silence_count = stop_silence;
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}
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s->stop_sample_count = 0;
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}
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if (s->stop_found_periods >= 0) {
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const int dst_pos = out_nb_samples * nb_channels;
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for (int ch = 0; ch < nb_channels; ch++)
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dst[dst_pos + ch] = stop[stop_pos + ch];
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out_nb_samples++;
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}
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*nb_out_samples = out_nb_samples;
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}
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